1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 # 3 # Use of this source code is governed by a BSD-style license 4 # that can be found in the LICENSE file in the root of the source 5 # tree. An additional intellectual property rights grant can be found 6 # in the file PATENTS. All contributing project authors may 7 # be found in the AUTHORS file in the root of the source tree. 8 9 import("../../build/webrtc.gni") 10 11 source_set("rtp_rtcp") { 12 sources = [ 13 "include/fec_receiver.h", 14 "include/receive_statistics.h", 15 "include/remote_ntp_time_estimator.h", 16 "include/rtp_header_parser.h", 17 "include/rtp_payload_registry.h", 18 "include/rtp_receiver.h", 19 "include/rtp_rtcp.h", 20 "include/rtp_rtcp_defines.h", 21 "mocks/mock_rtp_rtcp.h", 22 "source/bitrate.cc", 23 "source/bitrate.h", 24 "source/byte_io.h", 25 "source/dtmf_queue.cc", 26 "source/dtmf_queue.h", 27 "source/fec_private_tables_bursty.h", 28 "source/fec_private_tables_random.h", 29 "source/fec_receiver_impl.cc", 30 "source/fec_receiver_impl.h", 31 "source/forward_error_correction.cc", 32 "source/forward_error_correction.h", 33 "source/forward_error_correction_internal.cc", 34 "source/forward_error_correction_internal.h", 35 "source/h264_bitstream_parser.cc", 36 "source/h264_bitstream_parser.h", 37 "source/h264_sps_parser.cc", 38 "source/h264_sps_parser.h", 39 "source/mock/mock_rtp_payload_strategy.h", 40 "source/packet_loss_stats.cc", 41 "source/packet_loss_stats.h", 42 "source/producer_fec.cc", 43 "source/producer_fec.h", 44 "source/receive_statistics_impl.cc", 45 "source/receive_statistics_impl.h", 46 "source/remote_ntp_time_estimator.cc", 47 "source/rtcp_packet.cc", 48 "source/rtcp_packet.h", 49 "source/rtcp_packet/app.cc", 50 "source/rtcp_packet/app.h", 51 "source/rtcp_packet/bye.cc", 52 "source/rtcp_packet/bye.h", 53 "source/rtcp_packet/compound_packet.cc", 54 "source/rtcp_packet/compound_packet.h", 55 "source/rtcp_packet/dlrr.cc", 56 "source/rtcp_packet/dlrr.h", 57 "source/rtcp_packet/extended_jitter_report.cc", 58 "source/rtcp_packet/extended_jitter_report.h", 59 "source/rtcp_packet/nack.cc", 60 "source/rtcp_packet/nack.h", 61 "source/rtcp_packet/pli.cc", 62 "source/rtcp_packet/pli.h", 63 "source/rtcp_packet/psfb.cc", 64 "source/rtcp_packet/psfb.h", 65 "source/rtcp_packet/receiver_report.cc", 66 "source/rtcp_packet/receiver_report.h", 67 "source/rtcp_packet/report_block.cc", 68 "source/rtcp_packet/report_block.h", 69 "source/rtcp_packet/rrtr.cc", 70 "source/rtcp_packet/rrtr.h", 71 "source/rtcp_packet/rtpfb.cc", 72 "source/rtcp_packet/rtpfb.h", 73 "source/rtcp_packet/sli.cc", 74 "source/rtcp_packet/sli.h", 75 "source/rtcp_packet/tmmbn.cc", 76 "source/rtcp_packet/tmmbn.h", 77 "source/rtcp_packet/tmmbr.cc", 78 "source/rtcp_packet/tmmbr.h", 79 "source/rtcp_packet/transport_feedback.cc", 80 "source/rtcp_packet/transport_feedback.h", 81 "source/rtcp_packet/voip_metric.cc", 82 "source/rtcp_packet/voip_metric.h", 83 "source/rtcp_receiver.cc", 84 "source/rtcp_receiver.h", 85 "source/rtcp_receiver_help.cc", 86 "source/rtcp_receiver_help.h", 87 "source/rtcp_sender.cc", 88 "source/rtcp_sender.h", 89 "source/rtcp_utility.cc", 90 "source/rtcp_utility.h", 91 "source/rtp_format.cc", 92 "source/rtp_format.h", 93 "source/rtp_format_h264.cc", 94 "source/rtp_format_h264.h", 95 "source/rtp_format_video_generic.cc", 96 "source/rtp_format_video_generic.h", 97 "source/rtp_format_vp8.cc", 98 "source/rtp_format_vp8.h", 99 "source/rtp_format_vp9.cc", 100 "source/rtp_format_vp9.h", 101 "source/rtp_header_extension.cc", 102 "source/rtp_header_extension.h", 103 "source/rtp_header_parser.cc", 104 "source/rtp_packet_history.cc", 105 "source/rtp_packet_history.h", 106 "source/rtp_payload_registry.cc", 107 "source/rtp_receiver_audio.cc", 108 "source/rtp_receiver_audio.h", 109 "source/rtp_receiver_impl.cc", 110 "source/rtp_receiver_impl.h", 111 "source/rtp_receiver_strategy.cc", 112 "source/rtp_receiver_strategy.h", 113 "source/rtp_receiver_video.cc", 114 "source/rtp_receiver_video.h", 115 "source/rtp_rtcp_config.h", 116 "source/rtp_rtcp_impl.cc", 117 "source/rtp_rtcp_impl.h", 118 "source/rtp_sender.cc", 119 "source/rtp_sender.h", 120 "source/rtp_sender_audio.cc", 121 "source/rtp_sender_audio.h", 122 "source/rtp_sender_video.cc", 123 "source/rtp_sender_video.h", 124 "source/rtp_utility.cc", 125 "source/rtp_utility.h", 126 "source/ssrc_database.cc", 127 "source/ssrc_database.h", 128 "source/tmmbr_help.cc", 129 "source/tmmbr_help.h", 130 "source/video_codec_information.h", 131 "source/vp8_partition_aggregator.cc", 132 "source/vp8_partition_aggregator.h", 133 ] 134 135 configs += [ "../..:common_config" ] 136 public_configs = [ "../..:common_inherited_config" ] 137 138 if (is_clang) { 139 # Suppress warnings from Chrome's Clang plugins. 140 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. 141 configs -= [ "//build/config/clang:find_bad_constructs" ] 142 } 143 144 deps = [ 145 "../..:webrtc_common", 146 "../../system_wrappers", 147 "../remote_bitrate_estimator", 148 ] 149 150 if (is_win) { 151 cflags = [ 152 # TODO(jschuh): Bug 1348: fix this warning. 153 "/wd4267", # size_t to int truncations 154 155 # TODO(kjellander): Bug 261: fix this warning. 156 "/wd4373", # virtual function override. 157 ] 158 } 159 } 160
